The size of the output buffer is always known in advance and
the code has no alignment requirement (it uses mostly the PutBits API),
so allowing user-supplied buffers is trivial.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Given that the AVCodec.next pointer has now been removed, most of the
AVCodecs are not modified at all any more and can therefore be made
const (as this patch does); the only exceptions are the very few codecs
for external libraries that have a init_static_data callback.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Both AC-3 encoder share the same options, yet they are nevertheless
duplicated in the binary; and the options applying to the EAC-3 encoder
are a proper subset of the options for the AC-3 encoders, so that it can
use the same options as the former by putting the options specific to
AC-3 at the front. This commit implements this.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The AC3 encoder used to be a separate library called "Aften", which
got merged into libavcodec (literally, SVN commits and all).
The merge preserved as much features from the library as possible.
The code had two versions - a fixed point version and a floating
point version. FFmpeg had floating point DSP code used by other
codecs, the AC3 decoder including, so the floating-point DSP was
simply replaced with FFmpeg's own functions.
However, FFmpeg had no fixed-point audio code at that point. So
the encoder brought along its own fixed-point DSP functions,
including a fixed-point MDCT.
The fixed-point MDCT itself is trivially just a float MDCT with a
different type and each multiply being a fixed-point multiply.
So over time, it got refactored, and the FFT used for all other codecs
was templated.
Due to design decisions at the time, the fixed-point version of the
encoder operates at 16-bits of precision. Although convenient, this,
even at the time, was inadequate and inefficient. The encoder is noisy,
does not produce output comparable to the float encoder, and even
rings at higher frequencies due to the badly approximated winow function.
Enter MIPS (owned by Imagination Technologies at the time). They wanted
quick fixed-point decoding on their FPUless cores. So they contributed
patches to template the AC3 decoder so it had both a fixed-point
and a floating-point version. They also did the same for the AAC decoder.
They however, used 32-bit samples. Not 16-bits. And we did not have
32-bit fixed-point DSP functions, including an MDCT. But instead of
templating our MDCT to output 3 versions (float, 32-bit fixed and 16-bit fixed),
they simply copy-pasted their own MDCT into ours, and completely
ifdeffed our own MDCT code out if a 32-bit fixed point MDCT was selected.
This is also the status quo nowadays - 2 separate MDCTs, one which
produces floating point and 16-bit fixed point versions, and one
sort-of integrated which produces 32-bit MDCT.
MIPS weren't all that interested in encoding, so they left the encoder
as-is, and they didn't care much about the ifdeffery, mess or quality - it's
not their problem.
So the MDCT/FFT code has always been a thorn in anyone looking to clean up
code's eye.
Backstory over. Internally AC3 operates on 25-bit fixed-point coefficients.
So for the floating point version, the encoder simply runs the float MDCT,
and converts the resulting coefficients to 25-bit fixed-point, as AC3 is inherently
a fixed-point codec. For the fixed-point version, the input is 16-bit samples,
so to maximize precision the frame samples are analyzed and the highest set
bit is detected via ac3_max_msb_abs_int16(), and the coefficients are then
scaled up via ac3_lshift_int16(), so the input for the FFT is always at least 14 bits,
computed in normalize_samples(). After FFT, the coefficients are scaled up to 25 bits.
This patch simply changes the encoder to accept 32-bit samples, reusing
the already well-optimized 32-bit MDCT code, allowing us to clean up and drop
a large part of a very messy code of ours, as well as prepare for the future lavu/tx
conversion. The coefficients are simply scaled down to 25 bits during windowing,
skipping 2 separate scalings, as the hacks to extend precision are simply no longer
necessary. There's no point in running the MDCT always at 32 bits when you're
going to drop 6 bits off anyway, the headroom is plenty, and the MDCT rounds
properly.
This also makes the encoder even slightly more accurate over the float version,
as there's no coefficient conversion step necessary.
SIZE SAVINGS:
ARM32:
HARDCODED TABLES:
BASE - 10709590
DROP DSP - 10702872 - diff: -6.56KiB
DROP MDCT - 10667932 - diff: -34.12KiB - both: -40.68KiB
DROP FFT - 10336652 - diff: -323.52KiB - all: -364.20KiB
SOFTCODED TABLES:
BASE - 9685096
DROP DSP - 9678378 - diff: -6.56KiB
DROP MDCT - 9643466 - diff: -34.09KiB - both: -40.65KiB
DROP FFT - 9573918 - diff: -67.92KiB - all: -108.57KiB
ARM64:
HARDCODED TABLES:
BASE - 14641112
DROP DSP - 14633806 - diff: -7.13KiB
DROP MDCT - 14604812 - diff: -28.31KiB - both: -35.45KiB
DROP FFT - 14286826 - diff: -310.53KiB - all: -345.98KiB
SOFTCODED TABLES:
BASE - 13636238
DROP DSP - 13628932 - diff: -7.13KiB
DROP MDCT - 13599866 - diff: -28.38KiB - both: -35.52KiB
DROP FFT - 13542080 - diff: -56.43KiB - all: -91.95KiB
x86:
HARDCODED TABLES:
BASE - 12367336
DROP DSP - 12354698 - diff: -12.34KiB
DROP MDCT - 12331024 - diff: -23.12KiB - both: -35.46KiB
DROP FFT - 12029788 - diff: -294.18KiB - all: -329.64KiB
SOFTCODED TABLES:
BASE - 11358094
DROP DSP - 11345456 - diff: -12.34KiB
DROP MDCT - 11321742 - diff: -23.16KiB - both: -35.50KiB
DROP FFT - 11276946 - diff: -43.75KiB - all: -79.25KiB
PERFORMANCE (10min random s32le):
ARM32 - before - 39.9x - 0m15.046s
ARM32 - after - 28.2x - 0m21.525s
Speed: -30%
ARM64 - before - 36.1x - 0m16.637s
ARM64 - after - 36.0x - 0m16.727s
Speed: -0.5%
x86 - before - 184x - 0m3.277s
x86 - after - 190x - 0m3.187s
Speed: +3%
ff_eac3_exponent_init() set values twice when initializing a static
table; ergo the initialization code must not run concurrently with
a running EAC-3 encoder. Yet this code is executed every time an EAC-3
encoder is initialized. So use ff_thread_once() for this and also for a
similar initialization performed for all AC-3 encoders to make them all
init-threadsafe.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The AC-3 encoders (both floating- as well as fixed-point) as well as
the EAC-3 encoder share code: All use ff_ac3_encode_init() as well as
ff_ac3_encode_close(). Until ee726e777b
ff_ac3_encode_init() called ff_ac3_encode_close() to clean up on error.
Said commit removed this and instead set the FF_CODEC_CAP_INIT_CLEANUP
flag; but it did the latter only for the fixed-point AC-3 encoder and
not for the other two users of ff_ac3_encode_init(). This caused any
already allocated buffer to leak upon a subsequent error for the two
other encoders.
This commit fixes this by adding the FF_CODEC_CAP_INIT_CLEANUP flag
to the other two encoders using ff_ac3_encode_init().
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
* commit 'e22c63ac74b2968075be8bf0d2deb1ee63b28976':
ac3enc: Reshuffle some float/fixed-mode ifdefs to avoid a dummy function
Merged-by: James Almer <jamrial@gmail.com>
* commit '12004a9a7f20e44f4da2ee6c372d5e1794c8d6c5':
audiodsp/x86: yasmify vector_clipf_sse
audiodsp: reorder arguments for vector_clipf
Merged the version from Libav after a discussion with James Almer on
IRC:
19:22 <ubitux> jamrial: opinion on 12004a9a7f20e44f4da2ee6c372d5e1794c8d6c5?
19:23 <ubitux> it was apparently yasmified differently
19:23 <ubitux> (it depends on the previous commit arg shuffle)
19:24 <ubitux> i don't see the magic movsxdifnidn in your port btw
19:24 <ubitux> it's a port from 1d36defe94
19:25 <jamrial> seems better thanks to said arg shuffle
19:25 <jamrial> the loop is the same, but init is simpler
19:25 <jamrial> probably worth merging
19:25 <ubitux> OK
19:25 <ubitux> thanks
19:26 <jamrial> curious they didn't make len ptrdiff_t after the previous bunch of commits, heh
19:26 <ubitux> yeah indeed
Both commits are merged at the same time to prevent a conflict with our
existing yasmified ff_vector_clipf_sse.
Merged-by: Clément Bœsch <u@pkh.me>
* commit '9a9e2f1c8aa4539a261625145e5c1f46a8106ac2':
dsputil: Split audio operations off into a separate context
Conflicts:
configure
libavcodec/takdec.c
libavcodec/x86/Makefile
libavcodec/x86/dsputil.asm
libavcodec/x86/dsputil_init.c
libavcodec/x86/dsputil_mmx.c
libavcodec/x86/dsputil_x86.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '27631796c9d1b8146ad4a16e6539ecc08afa7565':
ac3: Only initialize float_dsp for the float encoder variant
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
wmaenc: use float planar sample format
(e)ac3enc: use planar sample format
aacenc: use planar sample format
adpcmenc: use planar sample format for adpcm_ima_wav and adpcm_ima_qt
adpcmenc: move 'ch' variable to higher scope
adpcmenc: fix 3 instances of variable shadowing
adpcm_ima_wav: simplify encoding
libvorbis: use planar sample format
libmp3lame: use planar sample formats
vorbisenc: use float planar sample format
ffm: do not write or read the audio sample format
parseutils: fix parsing of invalid alpha values
doc/RELEASE_NOTES: update for the 9 release.
smoothstreamingenc: Add a more verbose error message
smoothstreamingenc: Ignore the return value from mkdir
smoothstreamingenc: Try writing a manifest when opening the muxer
smoothstreamingenc: Move the output_chunk_list and write_manifest functions up
smoothstreamingenc: Properly return errors from ism_flush to the caller
smoothstreamingenc: Check the output UrlContext before accessing it
Conflicts:
doc/RELEASE_NOTES
libavcodec/aacenc.c
libavcodec/ac3enc_template.c
libavcodec/wmaenc.c
tests/ref/lavf/ffm
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
float_dsp: ppc: add a separate header for Altivec function prototypes
ARM: fix float_dsp breakage from d5a7229
Add a float DSP framework to libavutil
PPC: Move types_altivec.h and util_altivec.h from libavcodec to libavutil
ARM: Move asm.S from libavcodec to libavutil
vc1dsp: mark put/avg_vc1_mspel_mc() always_inline
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rtpdec_asf: Set the no_resync_search option for the chained asf demuxer
asfdec: Add an option for not searching for the packet markers
cosmetics: Clean up the tiffenc pix_fmts declaration to match the style of others
cosmetics: Align codec declarations
cosmetics: Convert mimic.c to utf-8
avconv: remove an unused function parameter.
avconv: remove now pointless variables.
avconv: drop support for building without libavfilter.
nellymoserenc: fix crash due to memsetting the wrong area.
libavformat: Only require first packet to be known for audio/video streams
avplay: Don't try to scale timestamps if the tb isn't set
Conflicts:
Changelog
configure
ffmpeg.c
libavcodec/aacenc.c
libavcodec/bmpenc.c
libavcodec/dnxhddec.c
libavcodec/dnxhdenc.c
libavcodec/ffv1.c
libavcodec/flacenc.c
libavcodec/fraps.c
libavcodec/huffyuv.c
libavcodec/libopenjpegdec.c
libavcodec/mpeg12enc.c
libavcodec/mpeg4videodec.c
libavcodec/pamenc.c
libavcodec/pgssubdec.c
libavcodec/pngenc.c
libavcodec/qtrleenc.c
libavcodec/rawdec.c
libavcodec/sgienc.c
libavcodec/tiffenc.c
libavcodec/v210dec.c
libavcodec/wmv2dec.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Also break some long lines, remove codec function placeholder comments
and add spaces in sample/pixel format lists.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master: (26 commits)
adxenc: use AVCodec.encode2()
adxenc: Use the AVFrame in ADXContext for coded_frame
indeo4: fix out-of-bounds function call.
configure: Restructure help output.
configure: Internal-only components should not be command-line selectable.
vorbisenc: use AVCodec.encode2()
libvorbis: use AVCodec.encode2()
libopencore-amrnbenc: use AVCodec.encode2()
ra144enc: use AVCodec.encode2()
nellymoserenc: use AVCodec.encode2()
roqaudioenc: use AVCodec.encode2()
libspeex: use AVCodec.encode2()
libvo_amrwbenc: use AVCodec.encode2()
libvo_aacenc: use AVCodec.encode2()
wmaenc: use AVCodec.encode2()
mpegaudioenc: use AVCodec.encode2()
libmp3lame: use AVCodec.encode2()
libgsmenc: use AVCodec.encode2()
libfaac: use AVCodec.encode2()
g726enc: use AVCodec.encode2()
...
Conflicts:
configure
libavcodec/Makefile
libavcodec/ac3enc.c
libavcodec/adxenc.c
libavcodec/libgsm.c
libavcodec/libvorbis.c
libavcodec/vorbisenc.c
libavcodec/wmaenc.c
tests/ref/acodec/g722
tests/ref/lavf/asf
tests/ref/lavf/ffm
tests/ref/lavf/mkv
tests/ref/lavf/mpg
tests/ref/lavf/rm
tests/ref/lavf/ts
tests/ref/seek/lavf_asf
tests/ref/seek/lavf_ffm
tests/ref/seek/lavf_mkv
tests/ref/seek/lavf_mpg
tests/ref/seek/lavf_rm
tests/ref/seek/lavf_ts
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (58 commits)
amrnbdec: check frame size before decoding.
cscd: use negative error values to indicate decode_init() failures.
h264: prevent overreads in intra PCM decoding.
FATE: do not decode audio in the nuv test.
dxa: set audio stream time base using the sample rate
psx-str: do not allow seeking by bytes
asfdec: Do not set AVCodecContext.frame_size
vqf: set packet parameters after av_new_packet()
mpegaudiodec: use DSPUtil.butterflies_float().
FATE: add mp3 test for sample that exhibited false overreads
fate: add cdxl test for bit line plane arrangement
vmnc: return error on decode_init() failure.
libvorbis: add/update error messages
libvorbis: use AVFifoBuffer for output packet buffer
libvorbis: remove unneeded e_o_s check
libvorbis: check return values for functions that can return errors
libvorbis: use float input instead of s16
libvorbis: do not flush libvorbis analysis if dsp state was not initialized
libvorbis: use VBR by default, with default quality of 3
libvorbis: fix use of minrate/maxrate AVOptions
...
Conflicts:
Changelog
doc/APIchanges
libavcodec/avcodec.h
libavcodec/dpxenc.c
libavcodec/libvorbis.c
libavcodec/vmnc.c
libavformat/asfdec.c
libavformat/id3v2enc.c
libavformat/internal.h
libavformat/mp3enc.c
libavformat/utils.c
libavformat/version.h
libswscale/utils.c
tests/fate/video.mak
tests/ref/fate/nuv
tests/ref/fate/prores-alpha
tests/ref/lavf/ffm
tests/ref/vsynth1/prores
tests/ref/vsynth2/prores
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (44 commits)
replacement Indeo 3 decoder
gsm demuxer: do not allocate packet twice.
flvenc: use first packet delay as global delay.
ac3enc: doxygen update.
imc: return error codes instead of 0 for error conditions.
imc: return meaningful error codes instead of -1
imc: do not set channel layout for stereo
imc: validate channel count
imc: check for ff_fft_init() failure
imc: check output buffer size before decoding
imc: use DSPContext.bswap16_buf() to byte-swap packet data
rtsp: add allowed_media_types option
libgsm: add flush function to reset the decoder state when seeking
libgsm: simplify decoding by using a loop
gsm: log error message when packet is too small
libgsmdec: do not needlessly set *data_size to 0
gsmdec: do not needlessly set *data_size to 0
gsmdec: add flush function to reset the decoder state when seeking
libgsmdec: check output buffer size before decoding
gsmdec: log error message when output buffer is too small.
...
Conflicts:
Changelog
ffplay.c
libavcodec/indeo3.c
libavcodec/mjpeg_parser.c
libavcodec/vp3.c
libavformat/cutils.c
libavformat/id3v2.c
libavutil/parseutils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Add some parameters to existing function documentation.
Remove some unneeded documentation.
Convert some static function documentation to non-doxygen style.
* qatar/master:
ac3enc: Add channel coupling support for the fixed-point AC-3 encoder.
ac3enc: scale floating-point coupling channel coefficients in scale_coefficients() rather than in apply_channel_coupling()
ac3enc: fix encoding of stereo ac3 files when rematrixing is disabled.
wavpack: fix wrong return value in wavpack_decode_block()
avconv: fix parsing metadata specifiers.
fate: use +frame+slice named constants instead of '3'
mpeg12: propagate more real return values through chunk decode error return and fix some indentation
wavpack: use context reset in appropriate places
avconv: move mux_preload and mux_max_delay to options context
avconv: move bitstream filters to options context.
avconv: move rate_emu to options context.
avconv: move max_frames to options context.
avconv: move metadata to options context.
avconv: move ts scale to options context.
avconv: move chapter maps to options context.
avconv: move metadata maps to options context.
avconv: move codec_names to options context.
Conflicts:
avconv.c
tests/fate-run.sh
Merged-by: Michael Niedermayer <michaelni@gmx.at>
